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    • 1. 发明申请
    • MULTI-CHANNEL AUDIO CODING
    • 多声道音频编码
    • WO2006103584A1
    • 2006-10-05
    • PCT/IB2006/050822
    • 2006-03-16
    • KONINKLIJKE PHILIPS ELECTRONICS N.V.CODING TECHNOLOGIES ABHOTHO, Gerard, H.BREEBAART, Dirk, J.SCHUIJERS, Erik, G., P.DEN BRINKER, Albertus, C.VILLEMOES, Lars, F.PURNHAGEN, HeikoRÖDÉN, Karl, J.
    • HOTHO, Gerard, H.BREEBAART, Dirk, J.SCHUIJERS, Erik, G., P.DEN BRINKER, Albertus, C.VILLEMOES, Lars, F.PURNHAGEN, HeikoRÖDÉN, Karl, J.
    • G10L19/00H04S3/00
    • G10L19/008H04R2420/07H04S3/008
    • A multi-channel audio encoder (10) for encoding a multi-channel audio signal (101), e.g. a 5.1 channel audio signal, into a spatial down-mix (102), e.g. a stereo signal, and associated parameters (104, 105). The encoder (10) comprises first and second units (110, 120). The first unit (110) encodes the multi-channel audio signal (101) into the spatial down-mix (102) and parameters (104). These parameters (104) enable a multi-channel decoder (20) to reconstruct the multi¬ channel audio signal (203) from the spatial down-mix (102). The second unit (120) generates, from the spatial down-mix (102), parameters (105) that enable the decoder to reconstruct the spatial down-mix (202) from an alternative down- mix (103), e.g. a so-called artistic down-mix that has been manually mixed in a sound studio. In this way, the decoder (20) can efficiently deal with a situation in which an alternative down-mix (103) is received instead of the regular spatial, down-mix (102). In the decoder (20), first the spatial down-mix (202) is reconstructed from the alternative down-mix (103) and the parameters (105). Next, the spatial down-mix (202) is decoded into the multi-channel audio signal (203).
    • 一种用于对多声道音频信号(101)进行编码的多声道音频编码器(10)。 5.1声道音频信号转换成空间缩混(102)。 立体声信号和相关参数(104,105)。 编码器(10)包括第一和第二单元(110,120)。 第一单元(110)将多声道音频信号(101)编码成空间缩混(102)和参数(104)。 这些参数(104)使得多声道解码器(20)能够从空间缩混(102)重建多声道音频信号(203)。 第二单元(120)从空间向下混合(102)产生能够使解码器从替代的下混合(103)重构空间缩混(202)的参数(105)。 所谓的艺术混合,已经在一个健全的工作室手工混合。 以这种方式,解码器(20)可以有效地处理其中接收替代的下混合(103)而不是常规的空间,下混合(102)的情况。 在解码器(20)中,首先,从替代的下混合(103)和参数(105)重构空间缩混(202)。 接下来,空间缩混(202)被解码为多声道音频信号(203)。
    • 2. 发明申请
    • MULTI-CHANNEL ENCODER
    • 多通道编码器
    • WO2005098824A1
    • 2005-10-20
    • PCT/IB2005/051040
    • 2005-03-25
    • KONINKLIJKE PHILIPS ELECTRONICS N.V.HOTHO, Gerard, H.BREEBAART, Dirk, J.VERBITSKIY, Evgeny, A.DEN BRINKER, Albertus, C.
    • HOTHO, Gerard, H.BREEBAART, Dirk, J.VERBITSKIY, Evgeny, A.DEN BRINKER, Albertus, C.
    • G10L19/02
    • G10L19/02G10L19/008G10L19/0204H04S3/008
    • There is described a method of encoding input signals (CHI to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data comprising down-mix output signals (610, 620) together with complementary parametric data (600). The method includes a first step of down-mixing input signals (CHI to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CHI to CH3; 400 to 450) during down-mixing to generate said parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CHI to CH3; 400 to 450) involves including information in the down­-mix signals (610, 620) which is useable during subsequent decoding of the down-mix output signals (610, 620) and the parametric data (600) to determine at least some parameter data and thereby enabling representations of the input signals (CHI to CH3; 400 to 450) to be subsequently regenerated. Coders for use in the encoder (5; 15) for performing essential signal processing operations therein are also elucidated.
    • 描述了在多通道编码器(5; 15)中对输入信号(CHI至CH3; 400至450)进行编码的方法,以生成包括下混合输出信号(610,620)的相应输出数据以及互补参数数据 (600)。 该方法包括将输入信号(CHI至CH3; 400至450)进行下混合以产生相应的下混合输出信号(610,620)的第一步骤,以及处理输入信号(CHI至CH3; 400至450),以产生与下混合输出信号(610,620)互补的所述参数数据(600)。 输入信号(CHI至CH3; 400至450)的处理涉及将下变频信号(610,620)中的信息包括在下混频输出信号(610,620)和参数数据的后续解码期间可用的信息 (600)以确定至少一些参数数据,从而使得能够随后再生所述输入信号(CHI至CH3; 400至450)的表示。 还阐明了用于编码器(5; 15)中用于执行其中的基本信号处理操作的编码器。
    • 8. 发明申请
    • AUDIO CODING
    • 音频编码
    • WO2005091275A1
    • 2005-09-29
    • PCT/IB2005/050847
    • 2005-03-08
    • KONINKLIJKE PHILIPS ELECTRONICS N.V.GERRITS, Andreas, J.DEN BRINKER, Albertus, C.
    • GERRITS, Andreas, J.DEN BRINKER, Albertus, C.
    • G10L19/08
    • G10L19/20G10L19/022G10L19/093
    • The method creates an audio stream comprising tracks of sinusoidal components linked across a plurality of sequential time segments. Segments in each track are weighted with a normal window (WI, W2, W3), and consecutive segments have a normal period of overlap (0) of their trailing edges and leading edges. Segments in which a transient5 component is determined are weighted with a first modified window (WIm) having a modified trailing edge, and the following segment in the track is weighted with a second modified window (W2m) having a modified leading edge, so that the modified trailing edge and the modified leading edge have a modified period of overlap (0m) that comprises the transient component and that is shorter than the normal period of overlap (0), and wherein the audio stream includes sinusoidal codes representing the frequency and the transient. According to the invention, the modified period of overlap (0m) depends on the frequency value (f).
    • 该方法创建包括跨多个连续时间段链接的正弦分量的轨道的音频流。 每个轨道中的段用正常窗口(WI,W2,W3)加权,并且连续的段具有其后沿和前沿的正常重叠周期(0)。 确定了瞬态5分量的分段用具有修改的后沿的第一修改窗口(WIm)加权,并且轨道中的下一分段被加权具有修改的前沿的第二修改窗口(W2m),使得 经修改的后沿和修改的前沿具有修改的重叠周期(0m),其包括瞬态分量并且短于正常重叠周期(0),并且其中音频流包括表示频率和瞬态的正弦代码 。 根据本发明,修改的重叠周期(0m)取决于频率值(f)。
    • 9. 发明申请
    • IMPROVING QUALITY OF DECODED AUDIO BY ADDING NOISE
    • 通过添加噪声来改善解码音质的质量
    • WO2005001814A1
    • 2005-01-06
    • PCT/IB2004/051010
    • 2004-06-25
    • KONINKLIJKE PHILIPS ELECTRONICS N.V.DEN BRINKER, Albertus, C.MYBURG, François, P.
    • DEN BRINKER, Albertus, C.MYBURG, François, P.
    • G10L21/02
    • G10L21/038
    • The present invention relates to a method of encoding and decoding an audio signal. The invention further relates to an arrangement for encoding and decoding an audio signal. The invention further relates to a computer-readable medium comprising a data record indicative of an audio signal and a device for communicating an audio signal having been encoded according to the present invention. By the method of encoding, a double description of the signal is obtained, where the encoding comprises two encoding steps, a first standard encoding and an additional second encoding. The second encoding is able to give a coarse description of the signal, such that a stochastic realization can be made and appropriate parts can be added to the decoded signal from the first decoding. The required description of the second encoder in order to make the realization of a stochastic signal possible requires a relatively low bit rate, while other double/multiple descriptions require a much higher bit rate.
    • 本发明涉及一种对音频信号进行编码和解码的方法。 本发明还涉及用于对音频信号进行编码和解码的装置。 本发明还涉及一种包括指示音频信号的数据记录的计算机可读介质和用于传送根据本发明已被编码的音频信号的设备。 通过编码的方法,获得对信号的双重描述,其中编码包括两个编码步骤,第一标准编码和附加的第二编码。 第二编码能够给出信号的粗略描述,使得可以进行随机实现,并且可以从第一解码将合适的部分添加到解码信号。 为了实现随机信号,第二编码器的所需描述可能需要相对低的比特率,而其他双重/多重描述需要高得多的比特率。
    • 10. 发明申请
    • AUDIO CODING
    • 音频编码
    • WO2003102922A1
    • 2003-12-11
    • PCT/IB2003/002044
    • 2003-05-16
    • KONINKLIJKE PHILIPS ELECTRONICS N.V.DEN BRINKER, Albertus, C.
    • DEN BRINKER, Albertus, C.
    • G10L19/04
    • G10L19/04G10L25/12
    • A method of encoding (14) an audio signal (x(n)) is disclosed. The method comprises the step of modelling (16) the audio signal in accordance with a frequency sensitizing parameter (() to provide a set of infinite impulse response (IIR) filter type characteristics ((0...k-1) of an order K and capable of being linearly combined with the sensitizing parameter (() to provide an estimate () for the audio signal (x(n)), the IIR type filter model satisfying the requirements of a minimum phase filter. The set of characteristics ((0...k-1) of order K are transformed as a function of the sensitizing parameter (() to provide a set of characteristics (c0...k) of order K+1 compatible with finite impulse response (FIR) filter type characteristics satisfying the requirements of a minimum phase filter. The set of characteristics (c0...k) of order K+1 are normalised to provide a set of characteristics (d1...k) of order K. An encoded audio stream (50) is generated to include representations (LAR,LSFs) of the normalised set of characteristics (d1...k) of order K.
    • 公开了一种编码(14)音频信号(x(n))的方法。 该方法包括根据频率敏化参数(())对音频信号进行建模(16)的步骤,以提供一组无限脉冲响应(IIR)滤波器类型特征((0 ... k-1) K,并且能够与敏化参数(())线性组合以提供音频信号(x(n))的估计(),满足最小相位滤波器要求的IIR型滤波器模型。 (0 ... k-1)作为敏化参数(()的函数被变换,以提供与有限脉冲响应(FIR)兼容的K + 1阶的特性(c0 ... k)集合, 滤波器类型特性满足最小相位滤波器的要求,对K + 1级的特性集(c0 ... k)进行归一化,以提供一个等级K的特征(d1 ... k)集合。编码音频 生成流(50)以包括秩K的特征(d1 ... k)的表示(LAR,LSF)。